Telecommunication systems such as the public switched telephone network (PSTN) and private branch exchanges (PBXs) are generally well known. The PSTN is now considered to be a digital system that is capable of carrying data at a theoretical speed of 64 kilobits per second (kbps). Despite many enhancements to the capacity, efficiency and performance that has undergone PSTN over the years, the voice quality is still limited to something less than “true voice” quality for several reasons. How the PSTN delivers voice from one telecommunication terminal to another is the culprit behind limited voice quality.
In transmitting voice from one telecommunications terminal to another several transformations take place. The caller's acoustic voice waves are converted to electrical analog signals by the microphone in the telephone handset of the near end telecommunications terminal which is connected to a central office in the caller's neighborhood through a subscriber line interface circuit. Latter performs duties such as powering the telecommunications terminal, detecting when the caller picks up or hangs up the receiver, and ringing the telecommunications terminal when required. A coder/decoder (codec) converts the analog voice signals to a digital data stream for easy routing through the network and delivery to the central office, located in the recipient's (far end) neighborhood, where the digital data stream is converted back into electrical analog signals. Then the handset speaker of the far end telecommunications terminal finally converts the analog signals to acoustic waves that are heard by the listener. The same process occurs in the opposite direction allowing the caller hearing the recipient voice.
One of the reasons the PSTN limits voice quality is to increase the call capacity of the network by reducing the data rate of each call. The PSTN confines each voice digital data stream to 64 kbit/s. This is achieved by sampling the voice signals at a rate of 8 kHz, and filtering out any frequencies less than 200 Hz and greater than 3.4 kHz. Amplitude compression is also used according to some so called μ-Law in the US or A-Law encoding in Europe resulting in an 8-bit, 8-kHz stream of data. This amplitude compression is part of a pulse code modulation (PCM) encoding techniques according to the ITU-T Recommendation G.711. Reversing this process at the receive end reproduces the caller's voice but without the original quality. This compression and expansion (companding) process of the G.711 algorithm adds distortion to the signal and gives a phone conversation its distinctive “low fidelity” quality. It is directly related to the used narrow bandwidth of about 3.5 kHz.
In lieu of PCM codecs, digital voice/speech codecs may be utilized by a telecommunication system to transmit audio signals in a different manner than the conventional PCM encoding techniques. Assuming that a suitable transmit bandwidth is available, such audio codecs can provide enhanced fidelity voice transmissions by incorporating audio characteristics such as tone, pitch, resonance, and the like, into the transmitted signal. For example, by leveraging the 64 kbps capability of current telephone networks, wideband voice codecs may be designed to provide high fidelity telephone calls in lieu of conventional audio calls that are governed by the PCM encoding protocols. Such high fidelity calls may be transmitted using a bandwidth that exceeds 3.5 kHz, e.g. 7 kHz or more, like defined already under ITU-T Recommendation G.711.
Due to the current standards that govern telecommunications systems, audio codecs may not be universally implemented in the many central offices associated with a given telecommunication system. Accordingly, an end-to-end high fidelity speech connection may not always be achieved if either of the respective central office do not utilize compatible audio codecs. Even if both ends (near and far end) support high fidelity speech communications, there must be a mechanism by which the central offices can communicate to determine whether (and which) wideband audio coding protocols are supported.
A possible signaling technique may simple employ a substantial portion of the normal operating bandwidth to transmit tones, or other signals at the beginning of a communication session. In US 2003/0224815 is described an example based on such technique. Although this procedure may effectively convey the necessary information between the central offices, the transmission of the signaling information may interfere with a call in progress and be noticeable to the end users.
In U.S. Pat. No. 6,353,666 is described an alternative for performing wide band communication sessions. It is time division multiplex (TDM) based, and therefore needs some specific in-band signaling which is proprietary. On FIG. 1 is shown the way a wideband communication session will be set up according to that prior art. At first, the near end telecommunications terminal 1 will send some set up request using some protocol Q.931 to its neighborhood local switch 2. That local switch 2 will forward such set up via integrated services digital network (ISUP) to the neighborhood local switch 3 of the far end telecommunications terminal 4. This local switch 3 will send an alert using the protocol Q.931 to the far end telecommunications terminal. A connect command will be answered by the far end telecommunications terminal 4 to be forward to the near end telecommunications terminal via the ISUP. Then, the near end telecommunications terminal 1 will ask if the far end telecommunications terminal is able to apply a wideband alternative of the encoding technique. After receiving a positive response from the far end telecommunications terminal 4, a second set up will be started followed by a training session between both telecommunications terminals. Only then a wideband telecommunications session will be started based on a proprietary in-band signaling. Such solution has an obvious drawback that it implies to control the codecs of both telecommunications terminals since both must be able to apply the proprietary wideband signaling.